static void
gst_level_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLevel *filter = GST_LEVEL (object);
switch (prop_id) {
case PROP_POST_MESSAGES:
/* fall-through */
case PROP_MESSAGE:
filter->post_messages = g_value_get_boolean (value);
break;
case PROP_INTERVAL:
filter->interval = g_value_get_uint64 (value);
if (GST_AUDIO_INFO_RATE (&filter->info)) {
filter->interval_frames =
GST_CLOCK_TIME_TO_FRAMES (filter->interval,
GST_AUDIO_INFO_RATE (&filter->info));
}
break;
case PROP_PEAK_TTL:
filter->decay_peak_ttl =
gst_guint64_to_gdouble (g_value_get_uint64 (value));
break;
case PROP_PEAK_FALLOFF:
filter->decay_peak_falloff = g_value_get_double (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/**
* gst_audio_info_is_equal:
* @info: a #GstAudioInfo
* @other: a #GstAudioInfo
*
* Compares two #GstAudioInfo and returns whether they are equal or not
*
* Returns: %TRUE if @info and @other are equal, else %FALSE.
*
* Since: 1.2
*
*/
gboolean
gst_audio_info_is_equal (const GstAudioInfo * info, const GstAudioInfo * other)
{
if (info == other)
return TRUE;
if (info->finfo == NULL || other->finfo == NULL)
return FALSE;
if (GST_AUDIO_INFO_FORMAT (info) != GST_AUDIO_INFO_FORMAT (other))
return FALSE;
if (GST_AUDIO_INFO_FLAGS (info) != GST_AUDIO_INFO_FLAGS (other))
return FALSE;
if (GST_AUDIO_INFO_LAYOUT (info) != GST_AUDIO_INFO_LAYOUT (other))
return FALSE;
if (GST_AUDIO_INFO_RATE (info) != GST_AUDIO_INFO_RATE (other))
return FALSE;
if (GST_AUDIO_INFO_CHANNELS (info) != GST_AUDIO_INFO_CHANNELS (other))
return FALSE;
if (GST_AUDIO_INFO_CHANNELS (info) > 64)
return TRUE;
if (memcmp (info->position, other->position,
GST_AUDIO_INFO_CHANNELS (info) * sizeof (GstAudioChannelPosition)) !=
0)
return FALSE;
return TRUE;
}
static gboolean
gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstOpusEnc *enc;
enc = GST_OPUS_ENC (benc);
g_mutex_lock (enc->property_lock);
enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
enc->sample_rate = GST_AUDIO_INFO_RATE (info);
gst_opus_enc_setup_channel_mappings (enc, info);
GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
enc->sample_rate);
/* handle reconfigure */
if (enc->state) {
opus_multistream_encoder_destroy (enc->state);
enc->state = NULL;
}
if (!gst_opus_enc_setup (enc))
return FALSE;
enc->frame_samples = gst_opus_enc_get_frame_samples (enc);
/* feedback to base class */
gst_opus_enc_setup_base_class (enc, benc);
g_mutex_unlock (enc->property_lock);
return TRUE;
}
static void
gst_level_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLevel *filter = GST_LEVEL (object);
switch (prop_id) {
case PROP_POST_MESSAGES:
/* fall-through */
case PROP_MESSAGE:
filter->post_messages = g_value_get_boolean (value);
break;
case PROP_INTERVAL:
filter->interval = g_value_get_uint64 (value);
/* Not exactly thread-safe, but property does not advertise that it
* can be changed at runtime anyway */
if (GST_AUDIO_INFO_RATE (&filter->info)) {
gst_level_recalc_interval_frames (filter);
}
break;
case PROP_PEAK_TTL:
filter->decay_peak_ttl =
gst_guint64_to_gdouble (g_value_get_uint64 (value));
break;
case PROP_PEAK_FALLOFF:
filter->decay_peak_falloff = g_value_get_double (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_lv2_source_set_caps (GstBaseSrc * base, GstCaps * caps)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (base, "received invalid caps");
return FALSE;
}
GST_DEBUG_OBJECT (lv2, "negotiated to caps %" GST_PTR_FORMAT, caps);
lv2->info = info;
gst_base_src_set_blocksize (base,
GST_AUDIO_INFO_BPF (&info) * lv2->samples_per_buffer);
if (!gst_lv2_setup (&lv2->lv2, GST_AUDIO_INFO_RATE (&info)))
goto no_instance;
return TRUE;
no_instance:
{
GST_ERROR_OBJECT (lv2, "could not create instance");
return FALSE;
}
}
static gboolean
gst_audio_filter_template_setup (GstAudioFilter * filter,
const GstAudioInfo * info)
{
GstAudioFilterTemplate *filter_template;
GstAudioFormat fmt;
gint chans, rate;
filter_template = GST_AUDIO_FILTER_TEMPLATE (filter);
rate = GST_AUDIO_INFO_RATE (info);
chans = GST_AUDIO_INFO_CHANNELS (info);
fmt = GST_AUDIO_INFO_FORMAT (info);
GST_INFO_OBJECT (filter_template, "format %d (%s), rate %d, %d channels",
fmt, GST_AUDIO_INFO_NAME (info), rate, chans);
/* if any setup needs to be done (like memory allocated), do it here */
/* The audio filter base class also saves the audio info in
* GST_AUDIO_FILTER_INFO(filter) so it's automatically available
* later from there as well */
return TRUE;
}
/* seek to time, will be called when we operate in push mode. In pull mode we
* get the requested byte offset. */
static gboolean
gst_lv2_source_do_seek (GstBaseSrc * base, GstSegment * segment)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstClockTime time;
gint samplerate, bpf;
gint64 next_sample;
GST_DEBUG_OBJECT (lv2, "seeking %" GST_SEGMENT_FORMAT, segment);
time = segment->position;
lv2->reverse = (segment->rate < 0.0);
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* now move to the time indicated, don't seek to the sample *after* the time */
next_sample = gst_util_uint64_scale_int (time, samplerate, GST_SECOND);
lv2->next_byte = next_sample * bpf;
if (samplerate == 0)
lv2->next_time = 0;
else
lv2->next_time =
gst_util_uint64_scale_round (next_sample, GST_SECOND, samplerate);
GST_DEBUG_OBJECT (lv2, "seeking next_sample=%" G_GINT64_FORMAT
" next_time=%" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (lv2->next_time));
g_assert (lv2->next_time <= time);
lv2->next_sample = next_sample;
if (!lv2->reverse) {
if (GST_CLOCK_TIME_IS_VALID (segment->start)) {
segment->time = segment->start;
}
} else {
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
segment->time = segment->stop;
}
}
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
lv2->sample_stop =
gst_util_uint64_scale_round (time, samplerate, GST_SECOND);
lv2->check_seek_stop = TRUE;
} else {
lv2->check_seek_stop = FALSE;
}
lv2->eos_reached = FALSE;
return TRUE;
}
gboolean
gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec, pa_format_info ** f,
guint * channels)
{
pa_format_info *format;
pa_sample_format_t sf = PA_SAMPLE_INVALID;
GstAudioInfo *ainfo = &spec->info;
format = pa_format_info_new ();
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW
&& GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
format->encoding = PA_ENCODING_PCM;
sf = PA_SAMPLE_ULAW;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW
&& GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
format->encoding = PA_ENCODING_PCM;
sf = PA_SAMPLE_ALAW;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
format->encoding = PA_ENCODING_PCM;
if (!gstaudioformat_to_pasampleformat (GST_AUDIO_INFO_FORMAT (ainfo), &sf))
goto fail;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3) {
format->encoding = PA_ENCODING_AC3_IEC61937;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3) {
format->encoding = PA_ENCODING_EAC3_IEC61937;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS) {
format->encoding = PA_ENCODING_DTS_IEC61937;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG) {
format->encoding = PA_ENCODING_MPEG_IEC61937;
} else {
goto fail;
}
if (format->encoding == PA_ENCODING_PCM) {
pa_format_info_set_sample_format (format, sf);
pa_format_info_set_channels (format, GST_AUDIO_INFO_CHANNELS (ainfo));
}
pa_format_info_set_rate (format, GST_AUDIO_INFO_RATE (ainfo));
if (!pa_format_info_valid (format))
goto fail;
*f = format;
*channels = GST_AUDIO_INFO_CHANNELS (ainfo);
return TRUE;
fail:
if (format)
pa_format_info_free (format);
return FALSE;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_acquire (GstAudioRingBuffer * rb,
GstAudioRingBufferSpec * spec)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
HRESULT ret;
BMDAudioSampleType sample_depth;
GST_DEBUG_OBJECT (self->sink, "Acquire");
if (spec->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz,
sample_depth, 2, bmdAudioOutputStreamContinuous);
if (ret != S_OK) {
GST_WARNING_OBJECT (self->sink, "Failed to enable audio output 0x%08x",
ret);
return FALSE;
}
g_mutex_lock (&self->output->lock);
self->output->audio_enabled = TRUE;
if (self->output->start_scheduled_playback)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
ret =
self->output->
output->SetAudioCallback (new GStreamerAudioOutputCallback (self));
if (ret != S_OK) {
GST_WARNING_OBJECT (self->sink,
"Failed to set audio output callback 0x%08x", ret);
return FALSE;
}
spec->segsize =
(spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time;
// set latency to one more segment as we need some headroom
spec->seglatency = spec->segtotal + 1;
rb->size = spec->segtotal * spec->segsize;
rb->memory = (guint8 *) g_malloc0 (rb->size);
return TRUE;
}
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_iir_filter_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioIIRFilter *self = GST_AUDIO_IIR_FILTER (base);
gint new_rate = GST_AUDIO_INFO_RATE (info);
if (GST_AUDIO_FILTER_RATE (self) != new_rate) {
g_signal_emit (G_OBJECT (self),
gst_audio_iir_filter_signals[SIGNAL_RATE_CHANGED], 0, new_rate);
}
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info);
}
static gboolean
gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf)
{
guint overlap;
guint out_start;
gboolean filled;
guint blocksize;
blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
blocksize = MAX (1, blocksize);
/* Overlap => mix */
if (aagg->priv->offset < pad->priv->output_offset)
out_start = pad->priv->output_offset - aagg->priv->offset;
else
out_start = 0;
overlap = pad->priv->size - pad->priv->position;
if (overlap > blocksize - out_start)
overlap = blocksize - out_start;
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
/* skip gap buffer */
GST_LOG_OBJECT (pad, "skipping GAP buffer");
pad->priv->output_offset += pad->priv->size - pad->priv->position;
pad->priv->position = pad->priv->size;
gst_buffer_replace (&pad->priv->buffer, NULL);
return FALSE;
}
filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg,
pad, inbuf, pad->priv->position, outbuf, out_start, overlap);
if (filled)
GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP);
pad->priv->position += overlap;
pad->priv->output_offset += overlap;
if (pad->priv->position == pad->priv->size) {
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
return FALSE;
}
return TRUE;
}
static gboolean
gst_celt_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstCeltEnc *enc;
GstCaps *otherpadcaps;
enc = GST_CELT_ENC (benc);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->rate = GST_AUDIO_INFO_RATE (info);
/* handle reconfigure */
if (enc->state) {
celt_encoder_destroy (enc->state);
enc->state = NULL;
}
if (enc->mode) {
celt_mode_destroy (enc->mode);
enc->mode = NULL;
}
memset (&enc->header, 0, sizeof (enc->header));
otherpadcaps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0);
gst_structure_get_int (ps, "frame-size", &enc->frame_size);
}
gst_caps_unref (otherpadcaps);
}
if (enc->requested_frame_size > 0)
enc->frame_size = enc->requested_frame_size;
GST_DEBUG_OBJECT (enc, "channels=%d rate=%d frame-size=%d",
enc->channels, enc->rate, enc->frame_size);
if (!gst_celt_enc_setup (enc))
return FALSE;
/* feedback to base class */
gst_audio_encoder_set_latency (benc,
gst_celt_enc_get_latency (enc), gst_celt_enc_get_latency (enc));
gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static void
gst_freeverb_init_rev_model (GstFreeverb * filter)
{
gfloat srfactor = GST_AUDIO_INFO_RATE (&filter->info) / 44100.0f;
GstFreeverbPrivate *priv = filter->priv;
freeverb_revmodel_free (filter);
priv->gain = fixedgain;
freeverb_comb_setbuffer (&priv->combL[0], combtuningL1 * srfactor);
freeverb_comb_setbuffer (&priv->combR[0], combtuningR1 * srfactor);
freeverb_comb_setbuffer (&priv->combL[1], combtuningL2 * srfactor);
freeverb_comb_setbuffer (&priv->combR[1], combtuningR2 * srfactor);
freeverb_comb_setbuffer (&priv->combL[2], combtuningL3 * srfactor);
freeverb_comb_setbuffer (&priv->combR[2], combtuningR3 * srfactor);
freeverb_comb_setbuffer (&priv->combL[3], combtuningL4 * srfactor);
freeverb_comb_setbuffer (&priv->combR[3], combtuningR4 * srfactor);
freeverb_comb_setbuffer (&priv->combL[4], combtuningL5 * srfactor);
freeverb_comb_setbuffer (&priv->combR[4], combtuningR5 * srfactor);
freeverb_comb_setbuffer (&priv->combL[5], combtuningL6 * srfactor);
freeverb_comb_setbuffer (&priv->combR[5], combtuningR6 * srfactor);
freeverb_comb_setbuffer (&priv->combL[6], combtuningL7 * srfactor);
freeverb_comb_setbuffer (&priv->combR[6], combtuningR7 * srfactor);
freeverb_comb_setbuffer (&priv->combL[7], combtuningL8 * srfactor);
freeverb_comb_setbuffer (&priv->combR[7], combtuningR8 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[0], allpasstuningL1 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[0], allpasstuningR1 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[1], allpasstuningL2 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[1], allpasstuningR2 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[2], allpasstuningL3 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[2], allpasstuningR3 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[3], allpasstuningL4 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[3], allpasstuningR4 * srfactor);
/* clear buffers */
freeverb_revmodel_init (filter);
/* set default values */
freeverb_allpass_setfeedback (&priv->allpassL[0], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[0], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassL[1], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[1], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassL[2], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[2], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassL[3], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[3], 0.5f);
}
static GstBuffer *
gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
GST_OBJECT_LOCK (bpad);
buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
GST_OBJECT_UNLOCK (bpad);
return buffer;
}
static void
gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
{
GstPad *pad;
guint i;
for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) {
gchar *name = g_strdup_printf ("src_%u", i);
GstCaps *srccaps;
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info);
gint rate = GST_AUDIO_INFO_RATE (&self->audio_info);
GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO;
CopyStickyEventsData data;
/* Set channel position if we know it */
if (self->keep_positions)
position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, format, rate, 1, &position);
srccaps = gst_audio_info_to_caps (&info);
pad = gst_pad_new_from_static_template (&src_template, name);
g_free (name);
gst_pad_use_fixed_caps (pad);
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
gst_pad_set_active (pad, TRUE);
data.pad = pad;
data.caps = srccaps;
gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data);
if (data.caps)
gst_pad_set_caps (pad, data.caps);
gst_element_add_pad (GST_ELEMENT (self), pad);
self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
gst_caps_unref (srccaps);
}
gst_element_no_more_pads (GST_ELEMENT (self));
self->srcpads = g_list_reverse (self->srcpads);
}
static guint
gst_osx_audio_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf;
double latency;
guint samples;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (!gst_core_audio_get_samples_and_latency (osxbuf->core_audio,
GST_AUDIO_INFO_RATE (&buf->spec.info), &samples, &latency)) {
return 0;
}
GST_DEBUG_OBJECT (buf, "Got latency: %f seconds -> %d samples",
latency, samples);
return samples;
}
static void
pcm_config_from_spec (struct pcm_config *config,
const GstAudioRingBufferSpec * spec)
{
gint64 frames;
config->format = pcm_format_from_gst (GST_AUDIO_INFO_FORMAT (&spec->info));
config->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
config->rate = GST_AUDIO_INFO_RATE (&spec->info);
gst_audio_info_convert (&spec->info,
GST_FORMAT_TIME, spec->latency_time * GST_USECOND,
GST_FORMAT_DEFAULT /* frames */ , &frames);
config->period_size = frames;
config->period_count = spec->buffer_time / spec->latency_time;
}
static gboolean
gst_ladspa_sink_type_set_caps (GstBaseSink * base, GstCaps * caps)
{
GstLADSPASink *ladspa = GST_LADSPA_SINK (base);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (base, "received invalid caps");
return FALSE;
}
GST_DEBUG_OBJECT (ladspa, "negotiated to caps %" GST_PTR_FORMAT, caps);
ladspa->info = info;
return gst_ladspa_setup (&ladspa->ladspa, GST_AUDIO_INFO_RATE (&info));
}
static GstFlowReturn
gst_chromaprint_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstChromaprint *chromaprint = GST_CHROMAPRINT (trans);
GstAudioFilter *filter = GST_AUDIO_FILTER (trans);
GstMapInfo map_info;
guint nsamples;
gint rate, channels;
rate = GST_AUDIO_INFO_RATE (&filter->info);
channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
if (G_UNLIKELY (rate <= 0 || channels <= 0))
return GST_FLOW_NOT_NEGOTIATED;
if (!chromaprint->record)
return GST_FLOW_OK;
if (!gst_buffer_map (buf, &map_info, GST_MAP_READ))
return GST_FLOW_ERROR;
nsamples = map_info.size / (channels * 2);
if (nsamples == 0)
goto end;
if (chromaprint->nsamples == 0) {
chromaprint_start (chromaprint->context, rate, channels);
}
chromaprint->nsamples += nsamples;
chromaprint->duration = chromaprint->nsamples / rate;
chromaprint_feed (chromaprint->context, map_info.data,
map_info.size / sizeof (guint16));
if (chromaprint->duration >= chromaprint->max_duration
&& !chromaprint->fingerprint) {
gst_chromaprint_create_fingerprint (chromaprint);
}
end:
gst_buffer_unmap (buf, &map_info);
return GST_FLOW_OK;
}
static gboolean
gst_ladspa_source_type_set_caps (GstBaseSrc * base, GstCaps * caps)
{
GstLADSPASource *ladspa = GST_LADSPA_SOURCE (base);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (base, "received invalid caps");
return FALSE;
}
GST_DEBUG_OBJECT (ladspa, "negotiated to caps %" GST_PTR_FORMAT, caps);
ladspa->info = info;
gst_base_src_set_blocksize (base,
GST_AUDIO_INFO_BPF (&info) * ladspa->samples_per_buffer);
return gst_ladspa_setup (&ladspa->ladspa, GST_AUDIO_INFO_RATE (&info));
}
static gboolean
gst_vorbis_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstVorbisEnc *vorbisenc;
vorbisenc = GST_VORBISENC (enc);
vorbisenc->channels = GST_AUDIO_INFO_CHANNELS (info);
vorbisenc->frequency = GST_AUDIO_INFO_RATE (info);
/* if re-configured, we were drained and cleared already */
if (!gst_vorbis_enc_setup (vorbisenc))
return FALSE;
/* feedback to base class */
gst_audio_encoder_set_latency (enc,
gst_vorbis_enc_get_latency (vorbisenc),
gst_vorbis_enc_get_latency (vorbisenc));
return TRUE;
}
static gboolean
gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (benc);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->rate = GST_AUDIO_INFO_RATE (info);
/* handle reconfigure */
if (enc->state) {
speex_encoder_destroy (enc->state);
enc->state = NULL;
}
if (!gst_speex_enc_setup (enc))
return FALSE;
/* feedback to base class */
gst_audio_encoder_set_latency (benc,
gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc));
gst_audio_encoder_set_lookahead (benc, enc->lookahead);
if (enc->nframes == 0) {
/* as many frames as available input allows */
gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
gst_audio_encoder_set_frame_max (benc, 0);
} else {
/* exactly as many frames as configured */
gst_audio_encoder_set_frame_samples_min (benc,
enc->frame_size * enc->nframes);
gst_audio_encoder_set_frame_samples_max (benc,
enc->frame_size * enc->nframes);
gst_audio_encoder_set_frame_max (benc, 1);
}
return TRUE;
}
GstFlowReturn AudioFileReader::handleSample(GstAppSink* sink)
{
GstSample* sample = gst_app_sink_pull_sample(sink);
if (!sample)
return GST_FLOW_ERROR;
GstBuffer* buffer = gst_sample_get_buffer(sample);
if (!buffer) {
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
GstCaps* caps = gst_sample_get_caps(sample);
if (!caps) {
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
GstAudioInfo info;
gst_audio_info_from_caps(&info, caps);
int frames = GST_CLOCK_TIME_TO_FRAMES(GST_BUFFER_DURATION(buffer), GST_AUDIO_INFO_RATE(&info));
// Check the first audio channel. The buffer is supposed to store
// data of a single channel anyway.
switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
gst_buffer_list_add(m_frontLeftBuffers, gst_buffer_ref(buffer));
m_channelSize += frames;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
gst_buffer_list_add(m_frontRightBuffers, gst_buffer_ref(buffer));
break;
default:
break;
}
gst_sample_unref(sample);
return GST_FLOW_OK;
}
gboolean
gst_pulse_fill_sample_spec (GstAudioRingBufferSpec * spec, pa_sample_spec * ss)
{
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
if (!gstaudioformat_to_pasampleformat (GST_AUDIO_INFO_FORMAT (&spec->info),
&ss->format))
return FALSE;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW) {
ss->format = PA_SAMPLE_ULAW;
} else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW) {
ss->format = PA_SAMPLE_ALAW;
} else
return FALSE;
ss->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
ss->rate = GST_AUDIO_INFO_RATE (&spec->info);
if (!pa_sample_spec_valid (ss))
return FALSE;
return TRUE;
}
static gboolean
gst_ce_mp3_enc_set_src_caps (GstCeAudEnc * ceaudenc, GstAudioInfo * info,
GstCaps ** caps, GstBuffer ** codec_data)
{
GstCeMp3Enc *mp3enc = GST_CE_MP3ENC (ceaudenc);
const gchar *mpegversion = NULL;
gboolean ret = TRUE;
ITTIAM_MP3ENC_Params *params;
mp3enc->channels = GST_AUDIO_INFO_CHANNELS (info);
GST_INFO_OBJECT (mp3enc, "Set src channels to %i", mp3enc->channels);
mp3enc->rate = GST_AUDIO_INFO_RATE (info);
if (mp3enc->rate >= 16000 & mp3enc->rate <= 24000) {
GST_DEBUG_OBJECT (mp3enc, "Setting samples per frame to 576");
gst_ce_audenc_set_frame_samples (ceaudenc, 576, 576);
} else {
GST_DEBUG_OBJECT (mp3enc, "Setting samples per frame to 1152");
gst_ce_audenc_set_frame_samples (ceaudenc, 1152, 1152);
}
return ret;
}
static void
gst_level_recalc_interval_frames (GstLevel * level)
{
GstClockTime interval = level->interval;
guint sample_rate = GST_AUDIO_INFO_RATE (&level->info);
guint interval_frames;
interval_frames = GST_CLOCK_TIME_TO_FRAMES (interval, sample_rate);
if (interval_frames == 0) {
GST_WARNING_OBJECT (level, "interval %" GST_TIME_FORMAT " is too small, "
"should be at least %" GST_TIME_FORMAT " for sample rate %u",
GST_TIME_ARGS (interval),
GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (1, sample_rate)), sample_rate);
interval_frames = 1;
}
level->interval_frames = interval_frames;
GST_INFO_OBJECT (level, "interval_frames now %u for interval "
"%" GST_TIME_FORMAT " and sample rate %u", interval_frames,
GST_TIME_ARGS (interval), sample_rate);
}
static gboolean
gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (enc);
/* parameters already parsed for us */
amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
/* we do not really accept other input, but anyway ... */
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
}
if (amrnbenc->rate != 8000) {
g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
gst_caps_unref (copy);
/* report needs to base class: hand one frame at a time */
gst_audio_encoder_set_frame_samples_min (enc, 160);
gst_audio_encoder_set_frame_samples_max (enc, 160);
gst_audio_encoder_set_frame_max (enc, 1);
return TRUE;
}
static gboolean
gst_audio_segment_clip_set_caps (GstSegmentClip * base, GstCaps * caps)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
gboolean ret;
GstAudioInfo info;
gint rate, channels, width;
gst_audio_info_init (&info);
ret = gst_audio_info_from_caps (&info, caps);
if (ret) {
rate = GST_AUDIO_INFO_RATE (&info);
channels = GST_AUDIO_INFO_CHANNELS (&info);
width = GST_AUDIO_INFO_WIDTH (&info);
GST_DEBUG_OBJECT (self, "Configured: rate %d channels %d width %d",
rate, channels, width);
self->rate = rate;
self->framesize = (width / 8) * channels;
}
return ret;
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer description */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
/* Define buffer */
memset (&wfx, 0, sizeof (WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Ignored for WAVE_FORMAT_PCM. */
wfx.cbSize = 0;
if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
goto dodgy_width;
/* Set the buffer size to two seconds.
This should never reached.
*/
dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
GST_DEBUG_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
/* Init secondary buffer desciption */
memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
descSecondary.dwFlags = 0;
descSecondary.dwReserved = 0;
/* This is not primary buffer so have to set size */
descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
descSecondary.lpwfxFormat = &wfx;
/* Create buffer */
hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
if (hRes != DS_OK)
goto capture_buffer;
dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG ("latency time: %" G_GUINT64_FORMAT " - buffer time: %"
G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
/* Buffer-time should be always more than 2*latency */
if (spec->buffer_time < spec->latency_time * 2) {
spec->buffer_time = spec->latency_time * 2;
GST_WARNING ("buffer-time was less than latency");
}
/* Save the times */
dsoundsrc->buffer_time = spec->buffer_time;
dsoundsrc->latency_time = spec->latency_time;
dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
dsoundsrc->latency_time / 1000000.0;
spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
wfx.nAvgBytesPerSec);
/* just in case */
if (spec->segsize < 1)
spec->segsize = 1;
spec->segtotal = GST_AUDIO_INFO_BPF (&spec->info) * 8 *
(wfx.nAvgBytesPerSec / spec->segsize);
GST_DEBUG_OBJECT (asrc,
"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
spec->segtotal);
/* Not read anything yet */
dsoundsrc->current_circular_offset = 0;
GST_DEBUG_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec);
return TRUE;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", wfx.wBitsPerSample), (NULL));
return FALSE;
}
}
static void
gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self,
const GstAudioInfo * info)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
gdouble *kernel = NULL;
gint rate, channels;
len = self->kernel_length;
if (info) {
rate = GST_AUDIO_INFO_RATE (info);
channels = GST_AUDIO_INFO_CHANNELS (info);
} else {
rate = GST_AUDIO_FILTER_RATE (self);
channels = GST_AUDIO_FILTER_CHANNELS (self);
}
if (rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp cutoff frequency between 0 and the nyquist frequency */
self->cutoff = CLAMP (self->cutoff, 0.0, rate / 2);
GST_DEBUG ("gst_audio_wsinclimit_: initializing filter kernel of length %d "
"with cutoff %.2lf Hz "
"for mode %s",
len, self->cutoff,
(self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
/* fill the kernel */
w = 2 * G_PI * (self->cutoff / rate);
kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == (len - 1) / 2.0)
kernel[i] = w;
else
kernel[i] = sin (w * (i - (len - 1) / 2)) / (i - (len - 1) / 2.0);
/* windowing */
switch (self->window) {
case WINDOW_HAMMING:
kernel[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / (len - 1)));
break;
case WINDOW_BLACKMAN:
kernel[i] *= (0.42 - 0.5 * cos (2 * G_PI * i / (len - 1)) +
0.08 * cos (4 * G_PI * i / (len - 1)));
break;
case WINDOW_GAUSSIAN:
kernel[i] *= exp (-0.5 * POW2 (3.0 / len * (2 * i - (len - 1))));
break;
case WINDOW_COSINE:
kernel[i] *= cos (G_PI * i / (len - 1) - G_PI / 2);
break;
case WINDOW_HANN:
kernel[i] *= 0.5 * (1 - cos (2 * G_PI * i / (len - 1)));
break;
}
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
sum += kernel[i];
for (i = 0; i < len; ++i)
kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
kernel[i] = -kernel[i];
if (len % 2 == 1) {
kernel[(len - 1) / 2] += 1.0;
} else {
kernel[len / 2 - 1] += 0.5;
kernel[len / 2] += 0.5;
}
}
gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
kernel, self->kernel_length, (len - 1) / 2, info);
}
